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Transcript
CS 218 F 2003 Nov 3 lecture: Streaming video/audio Adaptive encoding (eg, layered encoding) TCP friendliness References: J. Padhye, V.Firoiu, D. Towsley, J. Kurose” Modeling TCP Throughput: a Simple Model and its Empirical Validation” Sigcomm 98 S. Floyd, J. Padhye, J.Widmer “Equation Based Congestion Control for Unicast Applications”, Sigcomm 2000 Rejaie et al “ RAP: end to end control for real time streams” Infocom 99 Tang et al “ RCS: Rate Control Scheme..for high bit error rates” Streaming Important and growing application due to reduction of storage costs, increase in high speed net access from homes, enhancements to caching and introduction of QoS in IP networks Audio/Video file is segmented and sent over either TCP or UDP, public segmentation protocol: Real-Time Protocol (RTP) Using a Streaming Server This gets us around HTTP, allows a choice of UDP vs. TCP and the application layer protocol can be better tailored to Streaming; many enhancements options are possible (see next slide) Options When Using a Streaming Server Use UDP, and Server sends at a rate (Compression and Transmission) appropriate for client; to reduce jitter, Player buffers initially for 2-5 seconds, then starts display Use TCP, and sender sends at maximum possible rate under TCP; retransmit when error is encountered; Player uses a much large buffer to smooth delivery rate of TCP Real-Time Protocol (RTP) Provides standard packet format for real-time application Typically runs over UDP Specifies header fields below Payload Type: 7 bits, providing 128 possible different types of encoding; eg PCM, MPEG2 video, etc. Sequence Number: 16 bits; used to detect packet loss Real-Time Protocol (RTP) Timestamp: 32 bytes; gives the sampling instant of the first audio/video byte in the packet; used to remove jitter introduced by the network Synchronization Source identifier (SSRC): 32 bits; an id for the source of a stream; assigned randomly by the source RTP Control Protocol (RTCP) Protocol specifies report of packets exchanged between sources and destinations of multimedia information Three reports are defined: Receiver reception, Sender, and Source description Reports contain statistics such as the number of packets sent, number of packets lost, inter-arrival jitter Used to modify sender transmission rates and for diagnostics purposes